Knowledge (XXG)

Undersampling

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be made in a short enough interval that it can represent the instantaneous value of the signal with the highest frequency. This means that in the FM radio example above, the sampling circuit must be able to capture a signal with a frequency of 108 MHz, not 43.2 MHz. Thus, the sampling frequency may be only a little bit greater than 43.2 MHz, but the input bandwidth of the system must be at least 108 MHz. Similarly, the accuracy of the sampling timing, or
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When undersampling a real-world signal, the sampling circuit must be fast enough to capture the highest signal frequency of interest. Theoretically, each sample should be taken during an infinitesimally short interval, but this is not practically feasible. Instead, the sampling of the signal should
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Fig 1: The top 2 graphs depict Fourier transforms of 2 different functions that produce the same results when sampled at a particular rate. The baseband function is sampled faster than its Nyquist rate, and the bandpass function is undersampled, effectively converting it to baseband. The lower
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On the other hand, reconstruction is not usually the goal with sampled IF or RF signals. Rather, the sample sequence can be treated as ordinary samples of the signal frequency-shifted to near baseband, and digital demodulation can proceed on that basis, recognizing the spectrum mirroring when
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Plot of sample rates (y axis) versus the upper edge frequency (x axis) for a band of width 1; grays areas are combinations that are "allowed" in the sense that no two frequencies in the band alias to same frequency. The darker gray areas correspond to undersampling with the maximum value of
1279: 668: 115:. When the aliases are mutually exclusive (spectrally), the original transform and the original continuous function, or a frequency-shifted version of it (if desired), can be recovered from the samples. The first and third graphs of Figure 1 depict a 907: 978: 569: 355: 756: 771:= 4, the FM band spectrum fits easily between 1.5 and 2.0 times the sampling rate, for a sampling rate near 56 MHz (multiples of the Nyquist frequency being 28, 56, 84, 112, etc.). See the illustrations at the right. 275: 958: 1295:
Further generalizations of undersampling for the case of signals with multiple bands are possible, and signals over multidimensional domains (space or space-time) and have been worked out in detail by
1176: 134:) (shaded blue) and its mirror image (shaded beige). The condition for a non-destructive sample rate is that the aliases of both bands do not overlap when shifted by all integer multiples of 582: 141:. The fourth graph depicts the spectral result of sampling at the same rate as the baseband function. The rate was chosen by finding the lowest rate that is an integer sub-multiple of 160:.  Consequently, the bandpass function has effectively been converted to baseband. All the other rates that avoid overlap are given by these more general criteria, where 1124:{\displaystyle \left(-{\frac {n}{2}}f_{\mathrm {s} },-{\frac {n-1}{2}}f_{\mathrm {s} }\right)\cup \left({\frac {n-1}{2}}f_{\mathrm {s} },{\frac {n}{2}}f_{\mathrm {s} }\right)} 830: 758:
and this is a scenario of undersampling. In this case, the signal spectrum fits between 2 and 2.5 times the sampling rate (higher than 86.4–88 MHz but lower than 108–110 MHz).
408:= 5) sampling. An anti-alias filter quite tight to the FM radio band is required, and there's not room for stations at nearby expansion channels such as 87.9 without aliasing. 1420: 1313: 1150: 468: 790:
If the sampling theorem is interpreted as requiring twice the highest frequency, then the required sampling rate would be assumed to be greater than the
284: 420:= 4) sampling, showing plenty of room for bandpass anti-aliasing filter transition bands. The baseband image is frequency-reversed in this case (even 80:
of the high-frequency signal. Such sampling is also known as bandpass sampling, harmonic sampling, IF sampling, and direct IF-to-digital conversion.
692: 1413: 1344: 195: 1458: 914: 1406: 1448: 1371: 397:). For the case of a given sampling frequency, simpler formulae for the constraints on the signal's spectral band are given below. 1622: 1603: 1544: 101: 367:
Important signals of this sort include a radio's intermediate-frequency (IF), radio-frequency (RF) signal, and the individual
1274:{\displaystyle n\operatorname {sinc} \left({\frac {nt}{T}}\right)-(n-1)\operatorname {sinc} \left({\frac {(n-1)t}{T}}\right)} 1361: 1489: 663:{\displaystyle 1\leq n\leq \lfloor 5.4\rfloor =\left\lfloor {108\ \mathrm {MHz} \over 20\ \mathrm {MHz} }\right\rfloor } 1585: 54: 782: 1565: 1165:
The corresponding interpolation function is the bandpass filter given by this difference of lowpass impulse responses
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When one undersamples a bandpass signal, the samples are indistinguishable from the samples of a low-frequency
1474: 1388: 778: 902:{\displaystyle \scriptstyle \left(-{\frac {1}{2}}f_{\mathrm {s} },{\frac {1}{2}}f_{\mathrm {s} }\right),} 122:
The second graph of Figure 1 depicts the frequency profile of a bandpass function occupying the band (
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216 MHz. While this does satisfy the last condition on the sampling rate, it is grossly oversampled.
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graphs indicate how identical spectral results are created by the aliases of the sampling process.
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and the reconstructive interpolation function, or lowpass filter impulse response, is  
804: 70: 785:, must be appropriate for the frequencies being sampled 108MHz, not the lower sample rate. 1134: 119:
spectrum before and after being sampled at a rate that completely separates the aliases.
1159:= 1 (except that where the intervals come together at 0 frequency, they can be closed). 564:{\displaystyle W=f_{H}-f_{L}=108\ \mathrm {MHz} -88\ \mathrm {MHz} =20\ \mathrm {MHz} } 1652: 1627: 1575: 412: 400: 1617: 1608: 350:{\displaystyle 1\leq n\leq \left\lfloor {\frac {f_{H}}{f_{H}-f_{L}}}\right\rfloor } 146: 66: 364:
for which the condition is satisfied leads to the lowest possible sampling rates.
108:. The frequency offset between adjacent aliases is the sampling-rate, denoted by 1505: 416:
Spectrum of the FM radio band (88–108 MHz) and its baseband alias under 56 MHz (
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Spectrum of the FM radio band (88–108 MHz) and its baseband alias under 44 MHz (
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As we have seen, the normal baseband condition for reversible sampling is that
1637: 751:{\displaystyle 43.2\ \mathrm {MHz} <f_{\mathrm {s} }<44\ \mathrm {MHz} } 1398: 449: 1593: 433: 116: 77: 58: 28: 382:> 1, then the conditions result in what is sometimes referred to as 270:{\displaystyle {\frac {2f_{H}}{n}}\leq f_{s}\leq {\frac {2f_{L}}{n-1}}} 970:) = 0 outside the union of open positive and negative frequency bands 953:{\displaystyle \scriptstyle \operatorname {sinc} \left(t/T\right).} 411: 399: 93: 27: 18: 1402: 767:
will also lead to a useful sampling rate. For example, using
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To accommodate undersampling, the bandpass condition is that
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In the US, FM radio operates on the frequency band from
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Simulation and Software Radio for Mobile Communications
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of real-valued functions are symmetrical around the 0
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which includes the normal baseband condition as case
1137: 981: 917: 833: 695: 585: 471: 287: 198: 1584: 1498: 1467: 1436: 689:= 5 gives the lowest sampling frequencies interval 1273: 1144: 1123: 952: 901: 750: 662: 563: 349: 269: 73:), but is still able to reconstruct the signal. 1314:Oversampling and undersampling in data analysis 1414: 8: 604: 598: 1421: 1407: 1399: 577:The sampling conditions are satisfied for 1240: 1193: 1178: 1141: 1136: 1109: 1108: 1094: 1084: 1083: 1061: 1041: 1040: 1018: 1005: 1004: 990: 980: 933: 916: 883: 882: 868: 858: 857: 843: 832: 737: 721: 720: 702: 694: 642: 623: 614: 584: 550: 530: 510: 495: 482: 470: 334: 321: 310: 304: 286: 247: 237: 228: 209: 199: 197: 436:to illustrate the idea of undersampling. 1325: 1360:Hiroshi Harada, Ramjee Prasad (2002). 1336:Mixed-signal and DSP design techniques 459:= 108 MHz. The bandwidth is given by 7: 799:Note that if a band is sampled with 1110: 1085: 1042: 1006: 884: 859: 744: 741: 738: 722: 709: 706: 703: 649: 646: 643: 630: 627: 624: 557: 554: 551: 537: 534: 531: 517: 514: 511: 16:Signal processing sample technique 14: 100:of the Fourier transform (called 37:in the equations of this section. 1459:Nyquist–Shannon sampling theorem 145:and also satisfies the baseband 1545:Discrete-time Fourier transform 781:of the sampler, frequently the 102:discrete-time Fourier transform 1255: 1243: 1227: 1215: 811:, instead of a lowpass filter. 96:axis. After sampling, only a 1: 1490:Statistical signal processing 1389:"Undersampling SODAR Signals" 783:analog-to-digital converter 1675: 1539:Discrete Fourier transform 1516:Matched Z-transform method 1309:Drizzle (image processing) 1131:for some positive integer 823:) = 0 outside the interval 1533:Discrete cosine transform 1430:Digital signal processing 53:is a technique where one 1566:Post's inversion formula 1480:Digital image processing 680:can be 1, 2, 3, 4, or 5. 1475:Audio signal processing 1339:. Newnes. p. 20. 1275: 1146: 1125: 954: 903: 752: 664: 565: 425: 409: 351: 271: 61:-filtered signal at a 38: 25: 1276: 1147: 1126: 955: 904: 753: 665: 566: 415: 403: 352: 272: 31: 22: 1599:Anti-aliasing filter 1528:Constant-Q transform 1511:Advanced z-transform 1333:Walt Kester (2003). 1177: 1135: 979: 915: 831: 809:anti-aliasing filter 807:is required for the 779:aperture uncertainty 693: 583: 469: 285: 196: 1145:{\displaystyle n\,} 1556:Integral transform 1551:Impulse invariance 1523:Bilinear transform 1271: 1142: 1121: 950: 949: 899: 898: 748: 660: 561: 426: 410: 347: 277:, for any integer 267: 98:periodic summation 90:Fourier transforms 39: 26: 1659:Signal processing 1646: 1645: 1571:Starred transform 1561:Laplace transform 1485:Speech processing 1454:Estimation theory 1346:978-0-7506-7611-3 1265: 1206: 1102: 1077: 1034: 998: 876: 851: 763:A lower value of 736: 701: 654: 641: 622: 549: 529: 509: 388:bandpass sampling 341: 265: 219: 156: > 2 147:Nyquist criterion 69:(twice the upper 51:bandpass sampling 43:signal processing 1666: 1444:Detection theory 1423: 1416: 1409: 1400: 1393: 1392: 1387:Angelo Ricotta. 1384: 1378: 1377: 1366:. Artech House. 1357: 1351: 1350: 1330: 1280: 1278: 1277: 1272: 1270: 1266: 1261: 1241: 1211: 1207: 1202: 1194: 1151: 1149: 1148: 1143: 1130: 1128: 1127: 1122: 1120: 1116: 1115: 1114: 1113: 1103: 1095: 1090: 1089: 1088: 1078: 1073: 1062: 1052: 1048: 1047: 1046: 1045: 1035: 1030: 1019: 1011: 1010: 1009: 999: 991: 959: 957: 956: 951: 945: 941: 937: 908: 906: 905: 900: 894: 890: 889: 888: 887: 877: 869: 864: 863: 862: 852: 844: 805:band-pass filter 757: 755: 754: 749: 747: 734: 727: 726: 725: 712: 699: 669: 667: 666: 661: 659: 655: 653: 652: 639: 634: 633: 620: 615: 570: 568: 567: 562: 560: 547: 540: 527: 520: 507: 500: 499: 487: 486: 356: 354: 353: 348: 346: 342: 340: 339: 338: 326: 325: 315: 314: 305: 276: 274: 273: 268: 266: 264: 253: 252: 251: 238: 233: 232: 220: 215: 214: 213: 200: 172:are replaced by 71:cutoff frequency 1674: 1673: 1669: 1668: 1667: 1665: 1664: 1663: 1649: 1648: 1647: 1642: 1580: 1494: 1463: 1449:Discrete signal 1432: 1427: 1397: 1396: 1386: 1385: 1381: 1374: 1359: 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9: 6: 4: 3: 2: 1671: 1660: 1657: 1656: 1654: 1639: 1636: 1634: 1633:Undersampling 1631: 1629: 1628:Sampling rate 1626: 1624: 1621: 1619: 1616: 1614: 1610: 1607: 1605: 1602: 1600: 1597: 1595: 1592: 1591: 1589: 1587: 1583: 1577: 1576:Zak transform 1574: 1572: 1569: 1567: 1564: 1562: 1559: 1557: 1554: 1552: 1549: 1546: 1543: 1540: 1537: 1534: 1531: 1529: 1526: 1524: 1521: 1517: 1514: 1512: 1509: 1508: 1507: 1504: 1503: 1501: 1497: 1491: 1488: 1486: 1483: 1481: 1478: 1476: 1473: 1472: 1470: 1466: 1460: 1457: 1455: 1452: 1450: 1447: 1445: 1442: 1441: 1439: 1435: 1431: 1424: 1419: 1417: 1412: 1410: 1405: 1404: 1401: 1390: 1383: 1380: 1375: 1373:1-58053-044-3 1369: 1365: 1364: 1356: 1353: 1348: 1342: 1338: 1337: 1329: 1326: 1319: 1315: 1312: 1310: 1307: 1306: 1302: 1300: 1298: 1297:Igor Kluvánek 1293: 1291: 1267: 1262: 1258: 1252: 1249: 1246: 1237: 1233: 1230: 1224: 1221: 1218: 1212: 1208: 1203: 1199: 1196: 1190: 1186: 1183: 1180: 1173: 1172: 1171: 1170: 1169: 1168: 1158: 1154: 1138: 1117: 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72: 68: 64: 60: 56: 52: 48: 47:undersampling 44: 36: 30: 21: 1632: 1623:Quantization 1618:Oversampling 1609:Nyquist rate 1604:Downsampling 1382: 1362: 1355: 1335: 1328: 1294: 1289: 1286: 1166: 1164: 1156: 967: 963: 961: 910: 824: 820: 816: 814: 800: 792:Nyquist rate 791: 768: 764: 686: 677: 453: 442: 429: 421: 417: 405: 391: 387: 383: 379: 377: 368: 366: 361: 360:The highest 359: 281:satisfying: 278: 187: 180: 173: 169: 165: 161: 157: 150: 142: 135: 131: 127: 123: 121: 109: 105: 87: 75: 67:Nyquist rate 50: 46: 40: 34: 1506:Z-transform 676:Therefore, 373:filter bank 84:Description 63:sample rate 1638:Upsampling 1499:Techniques 1468:Sub-fields 1320:References 685:The value 65:below its 1613:frequency 1292:is even. 1250:− 1234:⁡ 1222:− 1213:− 1187:⁡ 1068:− 1054:∪ 1025:− 1016:− 988:− 923:⁡ 841:− 605:⌋ 599:⌊ 596:≤ 590:≤ 522:− 489:− 432:Consider 328:− 298:≤ 292:≤ 259:− 235:≤ 222:≤ 1653:Category 1594:Aliasing 1586:Sampling 1303:See also 657:⌋ 612:⌊ 434:FM radio 430:Example: 369:channels 344:⌋ 302:⌊ 117:baseband 59:bandpass 827:  106:aliases 55:samples 1547:(DTFT) 1437:Theory 1370:  1343:  735:  700:  640:  621:  548:  528:  508:  1541:(DFT) 1535:(DCT) 448:= 88 371:of a 78:alias 1368:ISBN 1341:ISBN 1231:sinc 1184:sinc 920:sinc 729:< 714:< 697:43.2 452:to 179:and 164:and 88:The 618:108 602:5.4 505:108 450:MHz 378:If 149:: 49:or 41:In 1655:: 1611:/ 1299:. 732:44 637:20 545:20 525:88 424:). 386:, 375:. 126:, 94:Hz 57:a 45:, 1422:e 1415:t 1408:v 1391:. 1376:. 1349:. 1290:n 1281:. 1268:) 1263:T 1259:t 1256:) 1253:1 1247:n 1244:( 1238:( 1228:) 1225:1 1219:n 1216:( 1209:) 1204:T 1200:t 1197:n 1191:( 1181:n 1167:: 1157:n 1152:. 1139:n 1118:) 1111:s 1106:f 1100:2 1097:n 1092:, 1086:s 1081:f 1075:2 1071:1 1065:n 1058:( 1050:) 1043:s 1038:f 1032:2 1028:1 1022:n 1013:, 1007:s 1002:f 996:2 993:n 984:( 968:f 966:( 964:X 947:. 943:) 939:T 935:/ 931:t 927:( 896:, 892:) 885:s 880:f 874:2 871:1 866:, 860:s 855:f 849:2 846:1 837:( 825:: 821:f 819:( 817:X 801:n 769:n 765:n 745:z 742:H 739:M 723:s 718:f 710:z 707:H 704:M 687:n 678:n 650:z 647:H 644:M 631:z 628:H 625:M 608:= 593:n 587:1 558:z 555:H 552:M 542:= 538:z 535:H 532:M 518:z 515:H 512:M 502:= 497:L 493:f 484:H 480:f 476:= 473:W 456:H 454:f 445:L 443:f 422:n 418:n 406:n 394:H 392:f 380:n 362:n 336:L 332:f 323:H 319:f 312:H 308:f 295:n 289:1 279:n 262:1 256:n 249:L 245:f 241:2 230:s 226:f 217:n 211:H 207:f 203:2 188:: 183:H 181:f 176:L 174:f 170:B 168:+ 166:A 162:A 158:B 153:s 151:f 143:A 138:s 136:f 132:B 130:+ 128:A 124:A 112:s 110:f 35:n

Index



signal processing
samples
bandpass
sample rate
Nyquist rate
cutoff frequency
alias
Fourier transforms
Hz
periodic summation
discrete-time Fourier transform
baseband
Nyquist criterion
filter bank


FM radio
MHz
aperture uncertainty
analog-to-digital converter
band-pass filter
anti-aliasing filter
Igor Kluvánek
Drizzle (image processing)
Oversampling and undersampling in data analysis
Mixed-signal and DSP design techniques
ISBN
978-0-7506-7611-3

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